1899 lines
85 KiB
Plaintext
1899 lines
85 KiB
Plaintext
GStreamer 1.20 Release Notes
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GStreamer 1.20.0 was released on 3 February 2022.
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See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
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version of this document.
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Last updated: Wednesday 2 February 2022, 23:30 UTC (log)
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Introduction
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The GStreamer team is proud to announce a new major feature release in
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the stable 1.x API series of your favourite cross-platform multimedia
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framework!
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As always, this release is again packed with many new features, bug
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fixes and other improvements.
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Highlights
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- Development in GitLab was switched to a single git repository
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containing all the modules
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- GstPlay: new high-level playback library, replaces GstPlayer
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- WebM Alpha decoding support
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- Encoding profiles can now be tweaked with additional
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application-specified element properties
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- Compositor: multi-threaded video conversion and mixing
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- RTP header extensions: unified support in RTP depayloader and
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payloader base classes
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- SMPTE 2022-1 2-D Forward Error Correction support
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- Smart encoding (pass through) support for VP8, VP9, H.265 in
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encodebin and transcodebin
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- Runtime compatibility support for libsoup2 and libsoup3 (libsoup3
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support experimental)
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- Video decoder subframe support
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- Video decoder automatic packet-loss, data corruption, and keyframe
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request handling for RTP / WebRTC / RTSP
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- mp4 and Matroska muxers now support profile/level/resolution changes
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for H.264/H.265 input streams (i.e. codec data changing on the fly)
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- mp4 muxing mode that initially creates a fragmented mp4 which is
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converted to a regular mp4 on EOS
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- Audio support for the WebKit Port for Embedded (WPE) web page source
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element
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- CUDA based video color space convert and rescale elements and
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upload/download elements
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- NVIDIA memory:NVMM support for OpenGL glupload and gldownload
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elements
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- Many WebRTC improvements
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- The new VA-API plugin implementation fleshed out with more decoders
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and new postproc elements
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- AppSink API to retrieve events in addition to buffers and buffer
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lists
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- AppSrc gained more configuration options for the internal queue
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(leakiness, limits in buffers and time, getters to read current
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levels)
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- Updated Rust bindings and many new Rust plugins
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- Improved support for custom minimal GStreamer builds
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- Support build against FFmpeg 5.0
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- Linux Stateless CODEC support gained MPEG-2 and VP9
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- Windows Direct3D11/DXVA decoder gained AV1 and MPEG-2 support
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- Lots of new plugins, features, performance improvements and bug
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fixes
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Major new features and changes
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Noteworthy new features and API
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- gst_element_get_request_pad() has been deprecated in favour of the
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newly-added gst_element_request_pad_simple() which does the exact
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same thing but has a less confusing name that hopefully makes clear
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that the function request a new pad rather than just retrieves an
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already-existing request pad.
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Development in GitLab was switched to a single git repository containing all the modules
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The GStreamer multimedia framework is a set of libraries and plugins
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split into a number of distinct modules which are released independently
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and which have so far been developed in separate git repositories in
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freedesktop.org GitLab.
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In addition to these separate git repositories there was a gst-build
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module that would use the Meson build system’s subproject feature to
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download each individual module and then build everything in one go. It
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would also provide an uninstalled development environment that made it
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easy to work on GStreamer and use or test versions other than the
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system-installed GStreamer version.
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All of these modules have now (as of 28 September 2021) been merged into
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a single git repository (“Mono repository” or “monorepo”) which should
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simplify development workflows and continuous integration, especially
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where changes need to be made to multiple modules at once.
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This mono repository merge will primarily affect GStreamer developers
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and contributors and anyone who has workflows based on the GStreamer git
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repositories.
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The Rust bindings and Rust plugins modules have not been merged into the
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mono repository at this time because they follow a different release
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cycle.
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The mono repository lives in the existing GStreamer core git repository
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in GitLab in the new main branch and all future development will happen
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on this branch.
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Modules will continue to be released as separate tarballs.
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For more details, please see the GStreamer mono repository FAQ.
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GstPlay: new high-level playback library replacing GstPlayer
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- GstPlay is a new high-level playback library that replaces the older
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GstPlayer API. It is basically the same API as GstPlayer but
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refactored to use bus messages for application notifications instead
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of GObject signals. There is still a signal adapter object for those
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who prefer signals. Since the existing GstPlayer API is already in
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use in various applications, it didn’t seem like a good idea to
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break it entirely. Instead a new API was added, and it is expected
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that this new GstPlay API will be moved to gst-plugins-base in
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future.
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- The existing GstPlayer API is scheduled for deprecation and will be
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removed at some point in the future (e.g. in GStreamer 1.24), so
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application developers are urged to migrate to the new GstPlay API
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at their earliest convenience.
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WebM alpha decoding
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- Implement WebM alpha decoding (VP8/VP9 with alpha), which required
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support and additions in various places. This is supported both with
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software decoders and hardware-accelerated decoders.
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- VP8/VP9 don’t support alpha components natively in the codec, so the
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way this is implemented in WebM is by encoding the alpha plane with
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transparency data as a separate VP8/VP9 stream. Inside the WebM
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container (a variant of Matroska) this is coded as a single video
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track with the “normal” VP8/VP9 video data making up the main video
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data and each frame of video having an encoded alpha frame attached
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to it as extra data ("BlockAdditional").
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- matroskademux has been extended extract this per-frame alpha side
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data and attach it in form of a GstVideoCodecAlphaMeta to the
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regular video buffers. Note that this new meta is specific to this
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VP8/VP9 alpha support and can’t be used to just add alpha support to
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other codecs that don’t support it. Lastly, matroskademux also
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advertises the fact that the streams contain alpha in the caps.
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- The new codecalpha plugin contains various bits of infrastructure to
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support autoplugging and debugging:
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- codecalphademux splits out the alpha stream from the metas on
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the regular VP8/VP9 buffers
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- alphacombine takes two decoded raw video streams (one alpha, one
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the regular video) and combines it into a video stream with
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alpha
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- vp8alphadecodebin + vp9alphadecodebin are wrapper bins that use
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the regular vp8dec and vp9dec software decoders to decode
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regular and alpha streams and combine them again. To decodebin
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these look like regular decoders.
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- The V4L2 CODEC plugin has stateless VP8/VP9 decoders that can
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decode both alpha and non-alpha stream with a single decoder
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instance
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- A new AV12 video format was added which is basically NV12 with an
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alpha plane, which is more convenient for many hardware-accelerated
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decoders.
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- Watch Nicolas Dufresne’s LCA 2022 talk “Bringing WebM Alpha support
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to GStreamer” for all the details and a demo.
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RTP Header Extensions Base Class and Automatic Header Extension Handling in RTP Payloaders and Depayloaders
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- RTP Header Extensions are specified in RFC 5285 and provide a way to
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add small pieces of data to RTP packets in between the RTP header
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and the RTP payload. This is often used for per-frame metadata,
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extended timestamps or other application-specific extra data. There
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are several commonly-used extensions specified in various RFCs, but
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senders are free to put any kind of data in there, as long as sender
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and receiver both know what that data is. Receivers that don’t know
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about the header extensions will just skip the extra data without
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ever looking at it. These header extensions can often be combined
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with any kind of payload format, so may need to be supported by many
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RTP payloader and depayloader elements.
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- Inserting and extracting RTP header extension data has so far been a
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bit inconvenient in GStreamer: There are functions to add and
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retrieve RTP header extension data from RTP packets, but nothing
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works automatically, even for common extensions. People would have
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to do the insertion/extraction either in custom elements
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before/after the RTP payloader/depayloader, or inside pad probes,
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which isn’t very nice.
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- This release adds various pieces of new infrastructure for generic
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RTP header extension handling, as well as some implementations for
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common extensions:
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- GstRTPHeaderExtension is a new helper base class for reading and
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writing RTP header extensions. Nominally this subclasses
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GstElement, but only so these extensions are stored in the
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registry where they can be looked up by URI or name. They don’t
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have pads and don’t get added to the pipeline graph as an
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element.
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- "add-extension" and "clear-extension" action signals on RTP
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payloaders and depayloaders for manual extension management
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- The "request-extension" signal will be emitted if an extension
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is encountered that requires explicit mapping by the application
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- new "auto-header-extension" property on RTP payloaders and
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depayloaders for automatic handling of known header extensions.
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This is enabled by default. The extensions must be signalled via
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caps / SDP.
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- RTP header extension implementations:
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- rtphdrextclientaudiolevel: Client-to-Mixer Audio Level
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Indication (RFC 6464) (also see below)
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- rtphdrextcolorspace: Color Space extension, extends RTP
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packets with color space and high dynamic range (HDR)
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information
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- rtphdrexttwcc: Transport Wide Congestion Control support
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- gst_rtp_buffer_remove_extension_data() is a new helper function to
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remove an RTP header extension from an RTP buffer
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- The existing gst_rtp_buffer_set_extension_data() now also supports
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shrinking the extension data in size
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AppSink and AppSrc improvements
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- appsink: new API to pull events out of appsink in addition to
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buffers and buffer lists.
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There was previously no way for users to receive incoming events
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from appsink properly serialised with the data flow, even if they
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are serialised events. The reason for that is that the only way to
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intercept events was via a pad probe on the appsink sink pad, but
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there is also internal queuing inside of appsink, so it’s difficult
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to ascertain the right order of everything in all cases.
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There is now a new "new-serialized-event" signal which will be
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emitted when there’s a new event pending (just like the existing
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"new-sample" signal). The "emit-signals" property must be set to
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TRUE in order to activate this (but it’s also fine to just pull from
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the application thread without using the signals).
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gst_app_sink_pull_object() and gst_app_sink_try_pull_object() can be
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used to pull out either an event or a new sample carrying a buffer
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or buffer list, whatever is next in the queue.
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EOS events will be filtered and will not be returned. EOS handling
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can be done the usual way, same as with _pull_sample().
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- appsrc: allow configuration of internal queue limits in time and
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buffers and add leaky mode.
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There is internal queuing inside appsrc so the application thread
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can push data into the element which will then be picked up by the
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source element’s streaming thread and pushed into the pipeline from
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that streaming thread. This queue is unlimited by default and until
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now it was only possible to set a maximum size limit in bytes. When
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that byte limit is reached, the pushing thread (application thread)
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would be blocked until more space becomes available.
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A limit in bytes is not particularly useful for many use cases, so
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now it is possible to also configure limits in time and buffers
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using the new "max-time" and "max-buffers" properties. Of course
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there are also matching new read-only"current-level-buffers" and
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"current-level-time properties" properties to query the current fill
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level of the internal queue in time and buffers.
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And as if that wasn’t enough the internal queue can also be
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configured as leaky using the new "leaky-type" property. That way
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when the queue is full the application thread won’t be blocked when
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it tries to push in more data, but instead either the new buffer
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will be dropped or the oldest data in the queue will be dropped.
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Better string serialization of nested GstCaps and GstStructures
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- New string serialisation format for structs and caps that can handle
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nested structs and caps properly by using brackets to delimit nested
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items (e.g. some-struct, some-field=[nested-struct, nested=true]).
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Unlike the default format the new variant can also support more than
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one level of nesting. For backwards-compatibility reasons the old
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format is still output by default when serialising caps and structs
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using the existing API. The new functions gst_caps_serialize() and
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gst_structure_serialize() can be used to output strings in the new
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format.
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Convenience API for custom GstMetas
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- New convenience API to register and create custom GstMetas:
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gst_meta_register_custom() and gst_buffer_add_custom_meta(). Such
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custom meta is backed by a GstStructure and does not require that
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users of the API expose their GstMeta implementation as public API
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for other components to make use of it. In addition, it provides a
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simpler interface by ignoring the impl vs. api distinction that the
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regular API exposes. This new API is meant to be the meta
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counterpart to custom events and messages, and to be more convenient
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than the lower-level API when the absolute best performance isn’t a
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requirement. The reason it’s less performant than a “proper” meta is
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that a proper meta is just a C struct in the end whereas this goes
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through the GstStructure API which has a bit more overhead, which
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for most scenarios is negligible however. This new API is useful for
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experimentation or proprietary metas, but also has some limitations:
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it can only be used if there’s a single producer of these metas;
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registering the same custom meta multiple times or from multiple
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places is not allowed.
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Additional Element Properties on Encoding Profiles
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- GstEncodingProfile: The new "element-properties" and
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gst_encoding_profile_set_element_properties() API allows
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applications to set additional element properties on encoding
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profiles to configure muxers and encoders. So far the encoding
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profile template was the only place where this could be specified,
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but often what applications want to do is take a ready-made encoding
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profile shipped by GStreamer or the application and then tweak the
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settings on top of that, which is now possible with this API. Since
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applications can’t always know in advance what encoder element will
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be used in the end, it’s even possible to specify properties on a
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per-element basis.
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Encoding Profiles are used in the encodebin, transcodebin and
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camerabin elements and APIs to configure output formats (containers
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and elementary streams).
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Audio Level Indication Meta for RFC 6464
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- New GstAudioLevelMeta containing Audio Level Indication as per RFC
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6464
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- The level element has been updated to add GstAudioLevelMeta on
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buffers if the "audio-level-meta" property is set to TRUE. This can
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then in turn be picked up by RTP payloaders to signal the audio
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level to receivers through RTP header extensions (see above).
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- New Client-to-Mixer Audio Level Indication (RFC6464) RTP Header
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Extension which should be automatically created and used by RTP
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payloaders and depayloaders if their "auto-header-extension"
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property is enabled and if the extension is part of the RTP caps.
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Automatic packet loss, data corruption and keyframe request handling for video decoders
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- The GstVideoDecoder base class has gained various new APIs to
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automatically handle packet loss and data corruption better by
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default, especially in RTP, RTSP and WebRTC streaming scenarios, and
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to give subclasses more control about how they want to handle
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missing data:
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- Video decoder subclasses can mark output frames as corrupted via
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the new GST_VIDEO_CODEC_FRAME_FLAG_CORRUPTED flag
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- A new "discard-corrupted-frames" property allows applications to
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configure decoders so that corrupted frames are directly
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discarded instead of being forwarded inside the pipeline. This
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is a replacement for the "output-corrupt" property of the FFmpeg
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decoders.
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- RTP depayloaders can now signal to decoders that data is missing
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when sending GAP events for lost packets. GAP events can be sent
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for various reason in a GStreamer pipeline. Often they are just
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used to let downstream elements know that there isn’t a buffer
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available at the moment, so downstream elements can move on
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instead of waiting for one. They are also sent by RTP
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depayloaders in the case that packets are missing, however, and
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so far a decoder was not able to differentiate the two cases.
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This has been remedied now: GAP events can be decorated with
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gst_event_set_gap_flags() and GST_GAP_FLAG_MISSING_DATA to let
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decoders now what happened, and decoders can then use that in
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some cases to handle missing data better.
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- The GstVideoDecoder::handle_missing_data vfunc was added to
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inform subclasses about packet loss or missing data and let them
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handle it in their own way if they like.
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- gst_video_decoder_set_needs_sync_point() lets subclasses signal
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that they need the stream to start with a sync point. If
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enabled, the base class will discard all non-sync point frames
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in the beginning and after a flush and does not pass them to the
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subclass. Furthermore, if the first frame is not a sync point,
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the base class will try and request a sync frame from upstream
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by sending a force-key-unit event (see next items).
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- New "automatic-request-sync-points" and
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"automatic-request-sync-point-flags" properties to automatically
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request sync points when needed, e.g. on packet loss or if the
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first frame is not a keyframe. Applications may want to enable
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this on decoders operating in e.g. RTP/WebRTC/RTSP receiver
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pipelines.
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- The new "min-force-key-unit-interval" property can be used to
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ensure there’s a minimal interval between keyframe requests to
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upstream (and/or the sender) and we’re not flooding the sender
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with key unit requests.
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- gst_video_decoder_request_sync_point() allows subclasses to
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request a new sync point (e.g. if they choose to do their own
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missing data handling). This will still honour the
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"min-force-key-unit-interval" property if set.
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Improved support for custom minimal GStreamer builds
|
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- Element registration and registration of other plugin features
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inside plugin init functions has been improved in order to
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facilitate minimal custom GStreamer builds.
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- A number of new macros have been added to declare and create
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per-element and per-plugin feature register functions in all
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plugins, and then call those from the per-plugin plugin_init
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functions:
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- GST_ELEMENT_REGISTER_DEFINE,
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GST_DEVICE_PROVIDER_REGISTER_DEFINE,
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GST_DYNAMIC_TYPE_REGISTER_DEFINE, GST_TYPE_FIND_REGISTER_DEFINE
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for the actual registration call with GStreamer
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- GST_ELEMENT_REGISTER, GST_DEVICE_PROVIDER_REGISTER,
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GST_DYNAMIC_TYPE_REGISTER, GST_PLUGIN_STATIC_REGISTER,
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GST_TYPE_FIND_REGISTER to call the registration function defined
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by the REGISTER_DEFINE macro
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- GST_ELEMENT_REGISTER_DECLARE,
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GST_DEVICE_PROVIDER_REGISTER_DECLARE,
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GST_DYNAMIC_TYPE_REGISTER_DECLARE,
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GST_TYPE_FIND_REGISTER_DECLARE to declare the registration
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function defined by the REGISTER_DEFINE macro
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- and various variants for advanced use cases.
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- This means that applications can call the per-element and per-plugin
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feature registration functions for only the elements they need
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instead of registering plugins as a whole with all kinds of elements
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that may not be required (e.g. encoder and decoder instead of just
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decoder). In case of static linking all unused functions and their
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dependencies would be removed in this case by the linker, which
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helps minimise binary size for custom builds.
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- gst_init() will automatically call a gst_init_static_plugins()
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function if one exists.
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- See the GStreamer static build documentation and Stéphane’s blog
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post Generate a minimal GStreamer build, tailored to your needs for
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more details.
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New elements
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- New aesdec and aesenc elements for AES encryption and decryption in
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a custom format.
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- New encodebin2 element with dynamic/sometimes source pads in order
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to support the option of doing the muxing outside of encodebin,
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e.g. in combination with a splitmuxsink.
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- New fakeaudiosink and videocodectestsink elements for testing and
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debugging (see below for more details)
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||
- rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
|
||
audio codec
|
||
|
||
- rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
|
||
2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
|
||
post.
|
||
|
||
- isac: new plugin wrapping the Internet Speech Audio Codec reference
|
||
encoder and decoder from the WebRTC project.
|
||
|
||
- asio: plugin for Steinberg ASIO (Audio Streaming Input/Output) API
|
||
|
||
- gssrc, gssink: add source and sink for Google Cloud Storage
|
||
|
||
- onnx: new plugin to apply ONNX neural network models to video
|
||
|
||
- openaptx: aptX and aptX-HD codecs using libopenaptx (v0.2.0)
|
||
|
||
- qroverlay, debugqroverlay: new elements that allow overlaying data
|
||
on top of video in the form of a QR code
|
||
|
||
- cvtracker: new OpenCV-based tracker element
|
||
|
||
- av1parse, vp9parse: new parsers for AV1 and VP9 video
|
||
|
||
- va: work on the new VA-API plugin implementation for
|
||
hardware-accelerated video decoding and encoding has continued at
|
||
pace, with various new decoders and filters having joined the
|
||
initial vah264dec:
|
||
|
||
- vah265dec: VA-API H.265 decoder
|
||
- vavp8dec: VA-API VP8 decoder
|
||
- vavp9dec: VA-API VP9 decoder
|
||
- vaav1dec: VA-API AV1 decoder
|
||
- vampeg2dec: VA-API MPEG-2 decoder
|
||
- vadeinterlace: : VA-API deinterlace filter
|
||
- vapostproc: : VA-API postproc filter (color conversion,
|
||
resizing, cropping, color balance, video rotation, skin tone
|
||
enhancement, denoise, sharpen)
|
||
|
||
See Víctor’s blog post “GstVA in GStreamer 1.20” for more details
|
||
and what’s coming up next.
|
||
|
||
- vaapiav1dec: new AV1 decoder element (in gstreamer-vaapi)
|
||
|
||
- msdkav1dec: hardware-accelerated AV1 decoder using the Intel Media
|
||
SDK / oneVPL
|
||
|
||
- nvcodec plugin for NVIDIA NVCODEC API for hardware-accelerated video
|
||
encoding and decoding:
|
||
|
||
- cudaconvert, cudascale: new CUDA based video color space convert
|
||
and rescale elements
|
||
- cudaupload, cudadownload: new helper elements for memory
|
||
transfer between CUDA and system memory spaces
|
||
- nvvp8sldec, nvvp9sldec: new GstCodecs-based VP8/VP9 decoders
|
||
|
||
- Various new hardware-accelerated elements for Windows:
|
||
|
||
- d3d11screencapturesrc: new desktop capture element, including a
|
||
GstDeviceProvider implementation to enumerate/select target
|
||
monitors for capture.
|
||
- d3d11av1dec and d3d11mpeg2dec: AV1 and MPEG-2 decoders
|
||
- d3d11deinterlace: deinterlacing filter
|
||
- d3d11compositor: video composing element
|
||
- see Windows section below for more details
|
||
|
||
- new Rust plugins:
|
||
|
||
- audiornnoise: Removes noise from an audio stream
|
||
- awstranscribeparse: Parses AWS audio transcripts into timed text
|
||
buffers
|
||
- ccdetect: Detects if valid closed captions are present in a
|
||
closed captions stream
|
||
- cea608tojson: Converts CEA-608 Closed Captions to a JSON
|
||
representation
|
||
- cmafmux: CMAF fragmented mp4 muxer
|
||
- dashmp4mux: DASH fragmented mp4 muxer
|
||
- isofmp4mux: ISO fragmented mp4 muxer
|
||
- ebur128level: EBU R128 Loudness Level Measurement
|
||
- ffv1dec: FFV1 video decoder
|
||
- gtk4paintablesink: GTK4 video sink, which provides a
|
||
GdkPaintable that can be rendered in various widgets
|
||
- hlssink3: HTTP Live Streaming sink
|
||
- hrtfrender: Head-Related Transfer Function (HRTF) renderer
|
||
- hsvdetector: HSV colorspace detector
|
||
- hsvfilter: HSV colorspace filter
|
||
- jsongstenc: Wraps buffers containing any valid top-level JSON
|
||
structures into higher level JSON objects, and outputs those as
|
||
ndjson
|
||
- jsongstparse: Parses ndjson as output by jsongstenc
|
||
- jsontovtt: converts JSON to WebVTT subtitles
|
||
- regex: Applies regular expression operations on text
|
||
- roundedcorners: Adds rounded corners to video
|
||
- spotifyaudiosrc: Spotify source
|
||
- textahead: Display upcoming text buffers ahead (e.g. for
|
||
Karaoke)
|
||
- transcriberbin: passthrough bin that transcribes raw audio to
|
||
closed captions using awstranscriber and puts the captions as
|
||
metas onto the video
|
||
- tttojson: Converts timed text to a JSON representation
|
||
- uriplaylistbin: Playlist source bin
|
||
- webpdec-rs: WebP image decoder with animation support
|
||
|
||
- New plugin codecalpha with elements to assist with WebM Alpha
|
||
decoding
|
||
|
||
- codecalphademux: Split stream with GstVideoCodecAlphaMeta into
|
||
two streams
|
||
- alphacombine: Combine two raw video stream (I420 or NV12) as one
|
||
stream with alpha channel (A420 or AV12)
|
||
- vp8alphadecodebin: A bin to handle software decoding of VP8 with
|
||
alpha
|
||
- vp9alphadecodebin: A bin to handle software decoding of VP9 with
|
||
alpha
|
||
|
||
- New hardware accelerated elements for Linux:
|
||
|
||
- v4l2slmpeg2dec: Support for Linux Stateless MPEG-2 decoders
|
||
- v4l2slvp9dec: Support for Linux Stateless VP9 decoders
|
||
- v4l2slvp8alphadecodebin: Support HW accelerated VP8 with alpha
|
||
layer decoding
|
||
- v4l2slvp9alphadecodebin: Support HW accelerated VP9 with alpha
|
||
layer decoding
|
||
|
||
New element features and additions
|
||
|
||
- assrender: handle more font mime types; better interaction with
|
||
matroskademux for embedded fonts
|
||
|
||
- audiobuffersplit: Add support for specifying output buffer size in
|
||
bytes (not just duration)
|
||
|
||
- audiolatency: new "samplesperbuffer" property so users can configure
|
||
the number of samples per buffer. The default value is 240 samples
|
||
which is equivalent to 5ms latency with a sample rate of 48000,
|
||
which might be larger than actual buffer size of audio capture
|
||
device.
|
||
|
||
- audiomixer, audiointerleave, GstAudioAggregator: now keep a count of
|
||
samples that are dropped or processed as statistic and can be made
|
||
to post QoS messages on the bus whenever samples are dropped by
|
||
setting the "qos-messages" property on input pads.
|
||
|
||
- audiomixer, compositor: improved handling of new inputs added at
|
||
runtime. New API was added to the GstAggregator base class to allow
|
||
subclasses to opt into an aggregation mode where inactive pads are
|
||
ignored when processing input buffers
|
||
(gst_aggregator_set_ignore_inactive_pads(),
|
||
gst_aggregator_pad_is_inactive()). An “inactive pad” in this context
|
||
is a pad which, in live mode, hasn’t yet received a first buffer,
|
||
but has been waited on at least once. What would happen usually in
|
||
this case is that the aggregator would wait for data on this pad
|
||
every time, up to the maximum configured latency. This would
|
||
inadvertently push mixer elements in live mode to the configured
|
||
latency envelope and delay processing when new inputs are added at
|
||
runtime until these inputs have actually produced data. This is
|
||
usually undesirable. With this new API, new inputs can be added
|
||
(requested) and configured and they won’t delay the data processing.
|
||
Applications can opt into this new behaviour by setting the
|
||
"ignore-inactive-pads" property on compositor, audiomixer or other
|
||
GstAudioAggregator-based elements.
|
||
|
||
- cccombiner: implement “scheduling” of captions. So far cccombiner’s
|
||
behaviour was essentially that of a funnel: it strictly looked at
|
||
input timestamps to associate together video and caption buffers.
|
||
Now it will try to smoothly schedule caption buffers in order to
|
||
have exactly one per output video buffer. This might involve
|
||
rewriting input captions, for example when the input is CDP then
|
||
sequence counters are rewritten, time codes are dropped and
|
||
potentially re-injected if the input video frame had a time code
|
||
meta. This can also lead to the input drifting from synchronisation,
|
||
when there isn’t enough padding in the input stream to catch up. In
|
||
that case the element will start dropping old caption buffers once
|
||
the number of buffers in its internal queue reaches a certain limit
|
||
(configurable via the "max-scheduled" property). The new original
|
||
funnel-like behaviour can be restored by setting the "scheduling"
|
||
property to FALSE.
|
||
|
||
- ccconverter: new "cdp-mode" property to specify which sections to
|
||
include in CDP packets (timecode, CC data, service info). Various
|
||
software, including FFmpeg’s Decklink support, fails parsing CDP
|
||
packets that contain anything but CC data in the CDP packets.
|
||
|
||
- clocksync: new "sync-to-first" property for automatic timestamp
|
||
offset setup: if set clocksync will set up the "ts-offset" value
|
||
based on the first buffer and the pipeline’s running time when the
|
||
first buffer arrived. The newly configured "ts-offset" in this case
|
||
would be the value that allows outputting the first buffer without
|
||
waiting on the clock. This is useful for example to feed a non-live
|
||
input into an already-running pipeline.
|
||
|
||
- compositor:
|
||
|
||
- multi-threaded input conversion and compositing. Set the
|
||
"max-threads" property to activate this.
|
||
- new "sizing-policy" property to support display aspect ratio
|
||
(DAR)-aware scaling. By default the image is scaled to fill the
|
||
configured destination rectangle without padding and without
|
||
keeping the aspect ratio. With sizing-policy=keep-aspect-ratio
|
||
the input image is scaled to fit the destination rectangle
|
||
specified by GstCompositorPad:{xpos, ypos, width, height}
|
||
properties preserving the aspect ratio. As a result, the image
|
||
will be centered in the destination rectangle with padding if
|
||
necessary.
|
||
- new "zero-size-is-unscaled" property on input pads. By default
|
||
pad width=0 or pad height=0 mean that the stream should not be
|
||
scaled in that dimension. But if the "zero-size-is-unscaled"
|
||
property is set to FALSE a width or height of 0 is instead
|
||
interpreted to mean that the input image on that pad should not
|
||
be composited, which is useful when creating animations where an
|
||
input image is made smaller and smaller until it disappears.
|
||
- improved handling of new inputs at runtime via
|
||
"ignore-inactive-pads"property (see above for details)
|
||
- allow output format with alpha even if none of the inputs have
|
||
alpha (also glvideomixer and other GstVideoAggregator
|
||
subclasses)
|
||
|
||
- dashsink: add H.265 codec support and signals for allowing custom
|
||
playlist/fragment output
|
||
|
||
- decodebin3:
|
||
|
||
- improved decoder selection, especially for hardware decoders
|
||
- make input activation “atomic” when adding inputs dynamically
|
||
- better interleave handling: take into account decoder latency
|
||
for interleave size
|
||
|
||
- decklink:
|
||
|
||
- Updated DeckLink SDK to 11.2 to support DeckLink 8K Pro
|
||
- decklinkvideosrc:
|
||
- More accurate and stable capture timestamps: use the
|
||
hardware reference clock time when the frame was finished
|
||
being captured instead of a clock time much further down the
|
||
road.
|
||
- Automatically detect widescreen vs. normal NTSC/PAL
|
||
|
||
- encodebin:
|
||
|
||
- add “smart encoding” support for H.265, VP8 and VP9 (i.e. only
|
||
re-encode where needed and otherwise pass through encoded video
|
||
as-is).
|
||
- H.264/H.265 smart encoding improvements: respect user-specified
|
||
stream-format, but if not specified default to avc3/hvc1 with
|
||
in-band SPS/PPS/VPS signalling for more flexibility.
|
||
- new encodebin2 element with dynamic/sometimes source pads in
|
||
order to support the option of doing the muxing outside of
|
||
encodebin, e.g. in combination with splitmuxsink.
|
||
- add APIs to set element properties on encoding profiles (see
|
||
below)
|
||
|
||
- errorignore: new "ignore-eos" property to also ignore FLOW_EOS from
|
||
downstream elements
|
||
|
||
- giosrc: add support for growing source files: applications can
|
||
specify that the underlying file being read is growing by setting
|
||
the "is-growing" property. If set, the source won’t EOS when it
|
||
reaches the end of the file, but will instead start monitoring it
|
||
and will start reading data again whenever a change is detected. The
|
||
new "waiting-data" and "done-waiting-data" signals keep the
|
||
application informed about the current state.
|
||
|
||
- gtksink, gtkglsink:
|
||
|
||
- scroll event support: forwarded as navigation events into the
|
||
pipeline
|
||
- "video-aspect-ratio-override" property to force a specific
|
||
aspect ratio
|
||
- "rotate-method" property and support automatic rotation based on
|
||
image tags
|
||
|
||
- identity: new "stats" property allows applications to retrieve the
|
||
number of bytes and buffers that have passed through so far.
|
||
|
||
- interlace: add support for more formats, esp 10-bit, 12-bit and
|
||
16-bit ones
|
||
|
||
- jack: new "low-latency" property for automatic latency-optimized
|
||
setting and "port-names" property to select ports explicitly
|
||
|
||
- jpegdec: support output conversion to RGB using libjpeg-turbo (for
|
||
certain input files)
|
||
|
||
- line21dec:
|
||
|
||
- "mode" property to control whether and how detected closed
|
||
captions should be inserted in the list of existing close
|
||
caption metas on the input frame (if any): add, drop, or
|
||
replace.
|
||
- "ntsc-only" property to only look for captions if video has NTSC
|
||
resolution
|
||
|
||
- line21enc: new "remove-caption-meta" to remove metas from output
|
||
buffers after encoding the captions into the video data; support for
|
||
CDP closed captions
|
||
|
||
- matroskademux, matroskamux: Add support for ffv1, a lossless
|
||
intra-frame video coding format.
|
||
|
||
- matroskamux: accept in-band SPS/PPS/VPS for H.264 and H.265
|
||
(i.e. stream-format avc3 and hev1) which allows on-the-fly
|
||
profile/level/resolution changes.
|
||
|
||
- matroskamux: new "cluster-timestamp-offset" property, useful for use
|
||
cases where the container timestamps should map to some absolute
|
||
wall clock time, for example.
|
||
|
||
- rtpsrc: add "caps" property to allow explicit setting of the caps
|
||
where needed
|
||
|
||
- mpegts: support SCTE-35 pass-through via new "send-scte35-events"
|
||
property on MPEG-TS demuxer tsdemux. When enabled, SCTE 35 sections
|
||
(e.g. ad placement opportunities) are forwarded as events downstream
|
||
where they can be picked up again by mpegtsmux. This required a
|
||
semantic change in the SCTE-35 section API: timestamps are now in
|
||
running time instead of muxer pts.
|
||
|
||
- tsdemux: Handle PCR-less MPEG-TS streams; more robust timestamp
|
||
handling in certain corner cases and for poorly muxed streams.
|
||
|
||
- mpegtsmux:
|
||
|
||
- More conformance improvements to make MPEG-TS analysers happy:
|
||
- PCR timing accuracy: Improvements to the way mpegtsmux
|
||
outputs PCR observations in CBR mode, so that a PCR
|
||
observation is always inserted when needed, so that we never
|
||
miss the configured pcr-interval, as that triggers various
|
||
MPEG-TS analyser errors.
|
||
- Improved PCR/SI scheduling
|
||
- Don’t write PCR until PAT/PMT are output to make sure streams
|
||
start cleanly with a PAT/PMT.
|
||
- Allow overriding the automatic PMT PID selection via
|
||
application-supplied PMT_%d fields in the prog-map
|
||
structure/property.
|
||
|
||
- mp4mux:
|
||
|
||
- new "first-moov-then-finalise" mode for fragmented output where
|
||
the output will start with a self-contained moov atom for the
|
||
first fragment, and then produce regular fragments. Then at the
|
||
end when the file is finalised, the initial moov is invalidated
|
||
and a new moov is written covering the entire file. This way the
|
||
file is a “fragmented mp4” file while it is still being written
|
||
out, and remains playable at all times, but at the end it is
|
||
turned into a regular mp4 file (with former fragment headers
|
||
remaining as unused junk data in the file).
|
||
- support H.264 avc3 and H.265 hvc1 stream formats as input where
|
||
the codec data is signalled in-band inside the bitstream instead
|
||
of caps/file headers.
|
||
- support profile/level/resolution changes for H.264/H.265 input
|
||
streams (i.e. codec data changing on the fly). Each codec_data
|
||
is put into its own SampleTableEntry inside the stsd, unless the
|
||
input is in avc3 stream format in which case it’s written
|
||
in-band and not in the headers.
|
||
|
||
- multifilesink: new ""min-keyframe-distance"" property to make
|
||
minimum distance between keyframes in next-file=key-frame mode
|
||
configurable instead of hard-coding it to 10 seconds.
|
||
|
||
- mxfdemux has seen a big refactoring to support non-frame wrappings
|
||
and more accurate timestamp/seek handling for some formats
|
||
|
||
- msdk plugin for hardware-accelerated video encoding and decoding
|
||
using the Intel Media SDK:
|
||
|
||
- oneVPL support (Intel oneAPI Video Processing Library)
|
||
- AV1 decoding support
|
||
- H.264 decoder now supports constrained-high and progressive-high
|
||
profiles
|
||
- H.264 encoder:
|
||
- more configuration options (properties):
|
||
"intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
|
||
"dblk-idc"
|
||
- H.265 encoder:
|
||
- can output main-still-picture profile
|
||
- now inserts HDR SEIs (mastering display colour volume and
|
||
content light level)
|
||
- more configuration options (properties):
|
||
"intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
|
||
"b-pyramid", "dblk-idc", "transform-skip"
|
||
- support for RGB 10bit format
|
||
- External bitrate control in encoders
|
||
- Video post proc element msdkvpp gained support for 12-bit pixel
|
||
formats P012_LE, Y212_LE and Y412_LE
|
||
|
||
- nvh264sldec: interlaced stream support
|
||
|
||
- openh264enc: support main, high, constrained-high and
|
||
progressive-high profiles
|
||
|
||
- openjpeg: support for multithreaded decoding and encoding
|
||
|
||
- rtspsrc: now supports IPv6 also for tunneled mode (RTSP-over-HTTP);
|
||
new "ignore-x-server-reply" property to ignore the
|
||
x-server-ip-address server header reply in case of HTTP tunneling,
|
||
as it is often broken.
|
||
|
||
- souphttpsrc: Runtime compatibility support for libsoup2 and
|
||
libsoup3. libsoup3 is the latest major version of libsoup, but
|
||
libsoup2 and libsoup3 can’t co-exist in the same process because
|
||
there is no namespacing or versioning for GObject types. As a
|
||
result, it would be awkward if the GStreamer souphttpsrc plugin
|
||
linked to a specific version of libsoup, because it would only work
|
||
with applications that use the same version of libsoup. To make this
|
||
work, the soup plugin now tries to determine the libsoup version
|
||
used by the application (and its other dependencies) at runtime on
|
||
systems where GStreamer is linked dynamically. libsoup3 support is
|
||
still considered somewhat experimental at this point. Distro
|
||
packagers please take note of the souphttpsrc plugin dependency
|
||
changes mentioned in the build and dependencies section below.
|
||
|
||
- srtsrc, srtsink: add signals for the application to accept/reject
|
||
incoming connections
|
||
|
||
- timeoverlay: new elapsed-running-time time mode which shows the
|
||
running time since the first running time (and each flush-stop).
|
||
|
||
- udpsrc: new timestamping mode to retrieve packet receive timestamps
|
||
from the kernel via socket control messages (SO_TIMESTAMPNS) on
|
||
supported platforms
|
||
|
||
- uritranscodebin: new setup-source and element-setup signals for
|
||
applications to configure elements used
|
||
|
||
- v4l2codecs plugin gained support for 4x4 and 32x32 tile formats
|
||
enabling some platforms or direct renders. Important memory usage
|
||
improvement.
|
||
|
||
- v4l2slh264dec now implements the final Linux uAPI as shipped on
|
||
Linux 5.11 and later.
|
||
|
||
- valve: add "drop-mode" property and provide two new modes of
|
||
operation: in drop-mode=forward-sticky-events sticky events
|
||
(stream-start, segment, tags, caps, etc.) are forwarded downstream
|
||
even when dropping is enabled; drop-mode=transform-to-gap will in
|
||
addition also convert buffers into gap events when dropping is
|
||
enabled, which lets downstream elements know that time is advancing
|
||
and might allow for preroll in many scenarios. By default all events
|
||
and all buffers are dropped when dropping is enabled, which can
|
||
cause problems with caps negotiation not progressing or branches not
|
||
prerolling when dropping is enabled.
|
||
|
||
- videocrop: support for many more pixel formats, e.g. planar YUV
|
||
formats with > 8bits and GBR* video formats; can now also accept
|
||
video not backed by system memory as long as downstream supports the
|
||
GstCropMeta
|
||
|
||
- videotestsrc: new smpte-rp-219 pattern for SMPTE75 RP-219 conformant
|
||
color bars
|
||
|
||
- vp8enc: finish support for temporal scalability: two new properties
|
||
("temporal-scalability-layer-flags",
|
||
"temporal-scalability-layer-sync-flags") and a unit change on the
|
||
"temporal-scalability-target-bitrate" property (now expects bps);
|
||
also make temporal scalability details available to RTP payloaders
|
||
as buffer metadata.
|
||
|
||
- vp9enc: new properties to tweak encoder performance:
|
||
|
||
- "aq-mode" to configure adaptive quantization modes
|
||
- "frame-parallel-decoding" to configure whether to create a
|
||
bitstream that reduces decoding dependencies between frames
|
||
which allows staged parallel processing of more than one video
|
||
frames in the decoder. (Defaults to TRUE)
|
||
- "row-mt", "tile-columns" and "tile-rows" so multithreading can
|
||
be enabled on a per-tile basis, instead of on a per tile-column
|
||
basis. In combination with the new "tile-rows" property, this
|
||
allows the encoder to make much better use of the available CPU
|
||
power.
|
||
|
||
- vp9dec, vp9enc: add support for 10-bit 4:2:0 and 4:2:2 YUV, as well
|
||
as 8-bit 4:4:4
|
||
|
||
- vp8enc, vp9enc now default to “good quality” for the deadline
|
||
property rather then “best quality”. Having the deadline set to best
|
||
quality causes the encoder to be absurdly slow, most real-life users
|
||
will prefer good-enough quality with better performance instead.
|
||
|
||
- wpesrc:
|
||
|
||
- implement audio support: a new sometimes source pad will be
|
||
created for each audio stream created by the web engine.
|
||
- move wpesrc to wpevideosrc and add a wrapper bin wpesrc to also
|
||
support audio
|
||
- also handles web:// URIs now (same as cefsrc)
|
||
- post messages with the estimated load progress on the bus
|
||
|
||
- x265enc: add negative DTS support, which means timestamps are now
|
||
offset by 1h same as with x264enc
|
||
|
||
RTP Payloaders and Depayloaders
|
||
|
||
- rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
|
||
audio codec
|
||
|
||
- rtph264depay:
|
||
|
||
- new "request-keyframe" property to make the depayloader
|
||
automatically request a new keyframe from the sender on packet
|
||
loss, consistent with the new property on rtpvp8depay.
|
||
- new "wait-for-keyframe" property to make depayloader wait for a
|
||
new keyframe at the beginning and after packet loss (only
|
||
effective if the depayloader outputs AUs), consistent with the
|
||
existing property on rtpvp8depay.
|
||
|
||
- rtpopuspay, rtpopusdepay: support libwebrtc-compatible multichannel
|
||
audio in addition to the previously supported multichannel audio
|
||
modes
|
||
|
||
- rtpopuspay: add DTX (Discontinuous Transmission) support
|
||
|
||
- rtpvp8depay: new "request-keyframe" property to make the depayloader
|
||
automatically request a new keyframe from the sender on packet loss.
|
||
|
||
- rtpvp8pay: temporal scaling support
|
||
|
||
- rtpvp9depay: Improved SVC handling (aggregate all layers)
|
||
|
||
RTP Infrastructure
|
||
|
||
- rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
|
||
2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
|
||
post.
|
||
|
||
- rtpreddec: BUNDLE support
|
||
|
||
- rtpredenc, rtpulpfecenc: add support for Transport-wide Congestion
|
||
Control (TWCC)
|
||
|
||
- rtpsession: new "twcc-feedback-interval" property to allow RTCP TWCC
|
||
reports to be scheduled on a timer instead of per marker-bit.
|
||
|
||
Plugin and library moves
|
||
|
||
- There were no plugin moves or library moves in this cycle.
|
||
|
||
Plugin removals
|
||
|
||
The following elements or plugins have been removed:
|
||
|
||
- The ofa audio fingerprinting plugin has been removed. The MusicIP
|
||
database has been defunct for years so this plugin is likely neither
|
||
useful nor used by anyone.
|
||
|
||
- The mms plugin containing mmssrc has been removed. It seems unlikely
|
||
anyone still needs this or that there are even any streams left out
|
||
there. The MMS protocol was deprecated in 2003 (in favour of RTSP)
|
||
and support for it was dropped with Microsoft Media Services 2008,
|
||
and Windows Media Player apparently also does not support it any
|
||
more.
|
||
|
||
Miscellaneous API additions
|
||
|
||
Core
|
||
|
||
- gst_buffer_new_memdup() is a convenience function for the
|
||
widely-used gst_buffer_new_wrapped(g_memdup(data,size),size)
|
||
pattern.
|
||
|
||
- gst_caps_features_new_single() creates a new single GstCapsFeatures,
|
||
avoiding the need to use the vararg function with NULL terminator
|
||
for simple cases.
|
||
|
||
- gst_element_type_set_skip_documentation() can be used by plugins to
|
||
signal that certain elements should not be included in the GStreamer
|
||
plugin documentation. This is useful for plugins where elements are
|
||
registered dynamically based on hardware capabilities and/or where
|
||
the available plugins and properties vary from system to system.
|
||
This is used in the d3d11 plugin for example to ensure that only the
|
||
list of default elements is advertised in the documentation.
|
||
|
||
- gst_type_find_suggest_empty_simple() is a new convenience function
|
||
for typefinders for cases where there’s only a media type and no
|
||
other fields.
|
||
|
||
- New API to create elements and set properties at construction time,
|
||
which is not only convenient, but also allows GStreamer elements to
|
||
have construct-only properties: gst_element_factory_make_full(),
|
||
gst_element_factory_make_valist(),
|
||
gst_element_factory_make_with_properties(),
|
||
gst_element_factory_create_full(),
|
||
gst_element_factory_create_valist(),
|
||
gst_element_factory_create_with_properties().
|
||
|
||
- GstSharedTaskPool: new “shared” task pool subclass with slightly
|
||
different default behaviour than the existing GstTaskPool which
|
||
would create unlimited number of threads for new tasks. The shared
|
||
task pool creates up to N threads (default: 1) and then distributes
|
||
pending tasks to those threads round-robin style, and blocks if no
|
||
thread is available. It is possible to join tasks. This can be used
|
||
by plugins to implement simple multi-threaded processing and is used
|
||
for the new multi-threaded video conversion and compositing done in
|
||
GstVideoAggregator, videoconverter and compositor.
|
||
|
||
Plugins Base Utils library
|
||
|
||
- GstDiscoverer:
|
||
|
||
- gst_discoverer_container_info_get_tags() was added to retrieve
|
||
global/container tags (vs. per-stream tags). Per-Stream tags can
|
||
be retrieved via the existing
|
||
gst_discoverer_stream_info_get_tags().
|
||
gst_discoverer_info_get_tags(), which for many files returns a
|
||
confusing mix of stream and container tags, has been deprecated
|
||
in favour of the container/stream-specific functions.
|
||
- gst_discoverer_stream_info_get_stream_number() returns a unique
|
||
integer identifier for a given stream within the given
|
||
GstDiscoverer context. (If this matches the stream number inside
|
||
the container bitstream that’s by coincidence and not by
|
||
design.)
|
||
|
||
- gst_pb_utils_get_caps_description_flags() can be used to query
|
||
whether certain caps represent a container, audio, video, image,
|
||
subtitles, tags, or something else. This only works for formats
|
||
known to GStreamer.
|
||
|
||
- gst_pb_utils_get_file_extension_from_caps() returns a possible file
|
||
extension for given caps.
|
||
|
||
- gst_codec_utils_h264_get_profile_flags_level(): Parses profile,
|
||
flags, and level from H.264 AvcC codec_data. The format of H.264
|
||
AVCC extradata/sequence_header is documented in the ITU-T H.264
|
||
specification section 7.3.2.1.1 as well as in ISO/IEC 14496-15
|
||
section 5.3.3.1.2.
|
||
|
||
- gst_codec_utils_caps_get_mime_codec() to convert caps to a RFC 6381
|
||
compatible MIME codec string codec. Useful for providing the codecs
|
||
field inside the Content-Type HTTP header for container formats,
|
||
such as mp4 or Matroska.
|
||
|
||
GStreamer OpenGL integration library and plugins
|
||
|
||
- glcolorconvert: added support for converting the video formats A420,
|
||
AV12, BGR, BGRA, RGBP and BGRP.
|
||
|
||
- Added support to GstGLBuffer for persistent buffer mappings where a
|
||
Pixel Buffer Object (PBO) can be mapped by both the CPU and the GPU.
|
||
This removes a memcpy() when uploading textures or vertices
|
||
particularly when software decoders (e.g. libav) are direct
|
||
rendering into our memory. Improves transfer performance
|
||
significantly. Requires OpenGL 4.4, GL_ARB_buffer_storage or
|
||
GL_EXT_buffer_storage
|
||
|
||
- Added various helper functions for handling 4x4 matrices of affine
|
||
transformations as used by GstVideoAffineTransformationMeta.
|
||
|
||
- Add support to GstGLContext for allowing the application to control
|
||
the config (EGLConfig, GLXConfig, etc) used when creating the OpenGL
|
||
context. This allows the ability to choose between RGB16 or RGB10A2
|
||
or RGBA8 back/front buffer configurations that were previously
|
||
hardcoded. GstGLContext also supports retrieving the configuration
|
||
it was created with or from an externally provide OpenGL context
|
||
handle. This infrastructure is also used to create a compatible
|
||
config from an application/externally provided OpenGL context in
|
||
order to improve compatibility with other OpenGL frameworks and GUI
|
||
toolkits. A new environment variable GST_GL_CONFIG was also added to
|
||
be able to request a specific configuration from the command line.
|
||
Note: different platforms will have different functionality
|
||
available.
|
||
|
||
- Add support for choosing between EGL and WGL at runtime when running
|
||
on Windows. Previously this was a build-time switch. Allows use in
|
||
e.g. Gtk applications on Windows that target EGL/ANGLE without
|
||
recompiling GStreamer. gst_gl_display_new_with_type() can be used by
|
||
applications to choose a specific display type to use.
|
||
|
||
- Build fixes to explicitly check for Broadcom-specific libraries on
|
||
older versions of the Raspberry Pi platform. The Broadcom OpenGL ES
|
||
and EGL libraries have different filenames. Using the vc4 Mesa
|
||
driver on the Raspberry Pi is not affected.
|
||
|
||
- Added support to glupload and gldownload for transferring RGBA
|
||
buffers using the memory:NVMM available on the Nvidia Tegra family
|
||
of embedded devices.
|
||
|
||
- Added support for choosing libOpenGL and libGLX as used in a GLVND
|
||
environment on unix-based platforms. This allows using desktop
|
||
OpenGL and EGL without pulling in any GLX symbols as would be
|
||
required with libGL.
|
||
|
||
Video library
|
||
|
||
- New raw video formats:
|
||
|
||
- AV12 (NV12 with alpha plane)
|
||
- RGBP and BGRP (planar RGB formats)
|
||
- ARGB64 variants with specified endianness instead of host
|
||
endianness:
|
||
- ARGB64_LE, ARGB64_BE
|
||
- RGBA64_BE, RGBA64_LE
|
||
- BGRA64_BE, BGRA64_LE
|
||
- ABGR64_BE, ABGR64_LE
|
||
|
||
- gst_video_orientation_from_tag() is new convenience API to parse the
|
||
image orientation from a GstTagList.
|
||
|
||
- GstVideoDecoder subframe support (see below)
|
||
|
||
- GstVideoCodecState now also carries some HDR metadata
|
||
|
||
- Ancillary video data: implement transform functions for AFD/Bar
|
||
metas, so they will be forwarded in more cases
|
||
|
||
MPEG-TS library
|
||
|
||
This library only handles section parsing and such, see above for
|
||
changes to the actual mpegtsmux and mpegtsdemux elements.
|
||
|
||
- many additions and improvements to SCTE-35 section parsing
|
||
- new API for fetching extended descriptors:
|
||
gst_mpegts_find_descriptor_with_extension()
|
||
- add support for SIT sections (Selection Information Tables)
|
||
- expose event-from-section constructor gst_event_new_mpegts_section()
|
||
- parse Audio Preselection Descriptor needed for Dolby AC-4
|
||
|
||
GstWebRTC library + webrtcbin
|
||
|
||
- Change the way in which sink pads and transceivers are matched
|
||
together to support easier usage. If a pad is created without a
|
||
specific index (i.e. using sink_%u as the pad template), then an
|
||
available compatible transceiver will be searched for. If a specific
|
||
index is requested (i.e. sink_1) then if a transceiver for that
|
||
m-line already exists, that transceiver must match the new sink pad
|
||
request. If there is no transceiver available in either scenario, a
|
||
new transceiver is created. If a mixture of both sink_1 and sink_%u
|
||
requests result in an impossible situation, an error will be
|
||
produced at pad request time or from create offer/answer.
|
||
|
||
- webrtcbin now uses regular ICE nomination instead of libnice’s
|
||
default of aggressive ICE nomination. Regular ICE nomination is the
|
||
default recommended by various relevant standards and improves
|
||
connectivity in specific network scenarios.
|
||
|
||
- Add support for limiting the port range used for RTP with the
|
||
addition of the min-rtp-port and max-rtp-port properties on the ICE
|
||
object.
|
||
|
||
- Expose the SCTP transport as a property on webrtcbin to more closely
|
||
match the WebRTC specification.
|
||
|
||
- Added support for taking into account the data channel transport
|
||
state when determining the value of the "connection-state" property.
|
||
Previous versions of the WebRTC spec did not include the data
|
||
channel state when computing this value.
|
||
|
||
- Add configuration for choosing the size of the underlying sockets
|
||
used for transporting media data
|
||
|
||
- Always advertise support for the transport-cc RTCP feedback protocol
|
||
as rtpbin supports it. For full support, the configured caps (input
|
||
or through codec-preferences) need to include the relevant RTP
|
||
header extension.
|
||
|
||
- Numerous fixes to caps and media handling to fail-fast when an
|
||
incompatible situation is detected.
|
||
|
||
- Improved support for attaching the required media after a remote
|
||
offer has been set.
|
||
|
||
- Add support for dynamically changing the amount of FEC used for a
|
||
particular stream.
|
||
|
||
- webrtcbin now stops further SDP processing at the first error it
|
||
encounters.
|
||
|
||
- Completed support for either local or the remote closing a data
|
||
channel.
|
||
|
||
- Various fixes when performing BUNDLEing of the media streams in
|
||
relation to RTX and FEC usage.
|
||
|
||
- Add support for writing out QoS DSCP marking on outgoing packets to
|
||
improve reliability in some network scenarios.
|
||
|
||
- Improvements to the statistics returned by the get-stats signal
|
||
including the addition of the raw statistics from the internal
|
||
RTPSource, the TWCC stats when available.
|
||
|
||
- The webrtc library does not expose any objects anymore with public
|
||
fields. Instead properties have been added to replace that
|
||
functionality. If you are accessing such fields in your application,
|
||
switch to the corresponding properties.
|
||
|
||
GstCodecs and Video Parsers
|
||
|
||
- Support for render delays to improve throughput across all CODECs
|
||
(used with NVDEC and V4L2).
|
||
- lots of improvements to parsers and the codec parsing decoder base
|
||
classes (H.264, H.265, VP8, VP9, AV1, MPEG-2) used for various
|
||
hardware-accelerated decoder APIs.
|
||
|
||
Bindings support
|
||
|
||
- gst_allocation_params_new() allocates a GstAllocationParams struct
|
||
on the heap. This should only be used by bindings (and freed via
|
||
gst_allocation_params_free() afterwards). In C code you would
|
||
allocate this on the stack and only init it in place.
|
||
|
||
- gst_debug_log_literal() can be used to log a string to the debug log
|
||
without going through any printf format expansion and associated
|
||
overhead. This is mostly useful for bindings such as the Rust
|
||
bindings which may have done their own formatting already .
|
||
|
||
- Provide non-inlined versions of refcounting APIs for various
|
||
GStreamer mini objects, so that they can be consumed by bindings
|
||
(e.g. gstreamer-sharp): gst_buffer_ref, gst_buffer_unref,
|
||
gst_clear_buffer, gst_buffer_copy, gst_buffer_replace,
|
||
gst_buffer_list_ref, gst_buffer_list_unref, gst_clear_buffer_list,
|
||
gst_buffer_list_copy, gst_buffer_list_replace, gst_buffer_list_take,
|
||
gst_caps_ref, gst_caps_unref, gst_clear_caps, gst_caps_replace,
|
||
gst_caps_take, gst_context_ref, gst_context_unref, gst_context_copy,
|
||
gst_context_replace, gst_event_replace, gst_event_steal,
|
||
gst_event_take, gst_event_ref, gst_event_unref, gst_clear_event,
|
||
gst_event_copy, gst_memory_ref, gst_memory_unref, gst_message_ref,
|
||
gst_message_unref, gst_clear_message, gst_message_copy,
|
||
gst_message_replace, gst_message_take, gst_promise_ref,
|
||
gst_promise_unref, gst_query_ref, gst_query_unref, gst_clear_query,
|
||
gst_query_copy, gst_query_replace, gst_query_take, gst_sample_ref,
|
||
gst_sample_unref, gst_sample_copy, gst_tag_list_ref,
|
||
gst_tag_list_unref, gst_clear_tag_list, gst_tag_list_replace,
|
||
gst_tag_list_take, gst_uri_copy, gst_uri_ref, gst_uri_unref,
|
||
gst_clear_uri.
|
||
|
||
- expose a GType for GstMiniObject
|
||
|
||
- gst_device_provider_probe() now returns non-floating device object
|
||
|
||
API Deprecations
|
||
|
||
- gst_element_get_request_pad() has been deprecated in favour of the
|
||
newly-added gst_element_request_pad_simple() which does the exact
|
||
same thing but has a less confusing name that hopefully makes clear
|
||
that the function request a new pad rather than just retrieves an
|
||
already-existing request pad.
|
||
|
||
- gst_discoverer_info_get_tags(), which for many files returns a
|
||
confusing mix of stream and container tags, has been deprecated in
|
||
favour of the container-specific and stream-specific functions,
|
||
gst_discoverer_container_info_get_tags() and
|
||
gst_discoverer_stream_info_get_tags().
|
||
|
||
- gst_video_sink_center_rect() was deprecated in favour of the more
|
||
generic newly-added gst_video_center_rect().
|
||
|
||
- The GST_MEMORY_FLAG_NO_SHARE flag has been deprecated, as it tends
|
||
to cause problems and prevents sub-buffering. If pooling or lifetime
|
||
tracking is required, memories should be allocated through a custom
|
||
GstAllocator instead of relying on the lifetime of the buffers the
|
||
memories were originally attached to, which is fragile anyway.
|
||
|
||
- The GstPlayer high-level playback library is being replaced with the
|
||
new GstPlay library (see above). GstPlayer should be considered
|
||
deprecated at this point and will be marked as such in the next
|
||
development cycle. Applications should be ported to GstPlay.
|
||
|
||
- Gstreamer Editing Services: ges_video_transition_set_border(),
|
||
ges_video_transition_get_border()
|
||
ges_video_transition_set_inverted()
|
||
ges_video_transition_is_inverted() have been deprecated, use
|
||
ges_timeline_element_set_children_properties() instead.
|
||
|
||
Miscellaneous performance, latency and memory optimisations
|
||
|
||
More video conversion fast paths
|
||
|
||
- v210 ↔ I420, YV12, Y42B, UYVY and YUY2
|
||
- A420 → RGB
|
||
|
||
Less jitter when waiting on the system clock
|
||
|
||
- Better system clock wait accuracy, less jitter: where available,
|
||
clock_nanosleep is used for higher accuracy for waits below 500
|
||
usecs, and waits below 2ms will first use the regular waiting system
|
||
and then clock_nanosleep for the remainder. The various wait
|
||
implementation have a latency ranging from 50 to 500+ microseconds.
|
||
While this is not a major issue when dealing with a low number of
|
||
waits per second (for ex: video), it does introduce a non-negligible
|
||
jitter for synchronisation of higher packet rate systems.
|
||
|
||
Video decoder subframe support
|
||
|
||
- The GstVideoDecoder base class gained API to process input at the
|
||
sub-frame level. That way video decoders can start decoding slices
|
||
before they have received the full input frame in its entirety (to
|
||
the extent this is supported by the codec, of course). This helps
|
||
with CPU utilisation and reduces latency.
|
||
|
||
- This functionality is now being used in the OpenJPEG JPEG 2000
|
||
decoder, the FFmpeg H.264 decoder (in case of NAL-aligned input) and
|
||
the OpenMAX H.264/H.265 decoders (in case of NAL-aligned input).
|
||
|
||
Miscellaneous other changes and enhancements
|
||
|
||
- GstDeviceMonitor no longer fails to start just because one of the
|
||
device providers failed to start. That could happen for example on
|
||
systems where the pulseaudio device provider is installed, but
|
||
pulseaudio isn’t actually running but ALSA is used for audio
|
||
instead. In the same vein the device monitor now keeps track of
|
||
which providers have been started (via the new
|
||
gst_device_provider_is_started()) and only stops actually running
|
||
device providers when stopping the device monitor.
|
||
|
||
- On embedded systems it can be useful to create a registry that can
|
||
be shared and read by multiple processes running as different users.
|
||
It is now possible to set the new GST_REGISTRY_MODE environment
|
||
variable to specify the file mode for the registry file, which by
|
||
default is set to be only user readable/writable.
|
||
|
||
- GstNetClientClock will signal lost sync in case the remote time
|
||
resets (e.g. because device power cycles), by emitting the “synced”
|
||
signal with synced=FALSE parameter, so applications can take action.
|
||
|
||
- gst_value_deserialize_with_pspec() allows deserialisation with a
|
||
hint for what the target GType should be. This allows for example
|
||
passing arrays of flags through the command line or
|
||
gst_util_set_object_arg(), eg: foo="<bar,bar+baz>".
|
||
|
||
- It’s now possible to create an empty GstVideoOverlayComposition
|
||
without any rectangles by passing a NULL rectangle to
|
||
gst_video_overlay_composition_new(). This is useful for bindings and
|
||
simplifies application code in some places.
|
||
|
||
Tracing framework, debugging and testing improvements
|
||
|
||
- New factories tracer to list loaded elements (and other plugin
|
||
features). This can be useful to collect a list of elements needed
|
||
for an application, which in turn can be used to create a tailored
|
||
minimal GStreamer build that contains just the elements needed and
|
||
nothing else.
|
||
- New plugin-feature-loaded tracing hook for use by tracers like the
|
||
new factories tracer
|
||
|
||
- GstHarness: Add gst_harness_set_live() so that harnesses can be set
|
||
to non-live and return is-live=false in latency queries if needed.
|
||
Default behaviour is to always return is-live=true in latency
|
||
queries.
|
||
|
||
- navseek: new "hold-eos" property. When enabled, the element will
|
||
hold back an EOS event until the next keystroke (via navigation
|
||
events). This can be used to keep a video sink showing the last
|
||
frame of a video pipeline until a key is pressed instead of tearing
|
||
it down immediately on EOS.
|
||
|
||
- New fakeaudiosink element: mimics an audio sink and can be used for
|
||
testing and CI pipelines on systems where no audio system is
|
||
installed or running. It differs from fakesink in that it only
|
||
support audio caps and syncs to the clock by default like a normal
|
||
audio sink. It also implements the GstStreamVolume interface like
|
||
most audio sinks do.
|
||
|
||
- New videocodectestsink element for video codec conformance testing:
|
||
Calculates MD5 checksums for video frames and skips any padding
|
||
whilst doing so. Can optionally also write back the video data with
|
||
padding removed into a file for easy byte-by-byte comparison with
|
||
reference data.
|
||
|
||
Tools
|
||
|
||
gst-inspect-1.0
|
||
|
||
- Can sort the list of plugins by passing --sort=name as command line
|
||
option
|
||
|
||
gst-launch-1.0
|
||
|
||
- will now error out on top-level properties that don’t exist and
|
||
which were silently ignored before
|
||
- On Windows the high-resolution clock is enabled now, which provides
|
||
better clock and timer performance on Windows (see Windows section
|
||
below for more details).
|
||
|
||
gst-play-1.0
|
||
|
||
- New --start-position command line argument to start playback from
|
||
the specified position
|
||
- Audio can be muted/unmuted in interactive mode by pressing the m
|
||
key.
|
||
- On Windows the high-resolution clock is enabled now (see Windows
|
||
section below for more details)
|
||
|
||
gst-device-monitor-1.0
|
||
|
||
- New --include-hidden command line argument to also show “hidden”
|
||
device providers
|
||
|
||
ges-launch-1.0
|
||
|
||
- New interactive mode that allows seeking and such. Can be disabled
|
||
by passing the --no-interactive argument on the command line.
|
||
- Option to forward tags
|
||
- Allow using an existing clip to determine the rendering format (both
|
||
topology and profile) via new --profile-from command line argument.
|
||
|
||
GStreamer RTSP server
|
||
|
||
- GstRTSPMediaFactory gained API to disable RTCP
|
||
(gst_rtsp_media_factory_set_enable_rtcp(), "enable-rtcp" property).
|
||
Previously RTCP was always allowed for all RTSP medias. With this
|
||
change it is possible to disable RTCP completely, irrespective of
|
||
whether the client wants to do RTCP or not.
|
||
|
||
- Make a mount point of / work correctly. While not allowed by the
|
||
RTSP 2 spec, the RTSP 1 spec is silent on this and it is used in the
|
||
wild. It is now possible to use / as a mount path in
|
||
gst-rtsp-server, e.g. rtsp://example.com/ would work with this now.
|
||
Note that query/fragment parts of the URI are not necessarily
|
||
correctly handled, and behaviour will differ between various
|
||
client/server implementations; so use it if you must but don’t bug
|
||
us if it doesn’t work with third party clients as you’d hoped.
|
||
|
||
- multithreading fixes (races, refcounting issues, deadlocks)
|
||
|
||
- ONVIF audio backchannel fixes
|
||
|
||
- ONVIF trick mode optimisations
|
||
|
||
- rtspclientsink: new "update-sdp" signal that allows updating the SDP
|
||
before sending it to the server via ANNOUNCE. This can be used to
|
||
add additional metadata to the SDP, for example. The order and
|
||
number of medias must not be changed, however.
|
||
|
||
GStreamer VAAPI
|
||
|
||
- new AV1 decoder element (vaapiav1dec)
|
||
|
||
- H.264 decoder: handle stereoscopic 3D video with frame packing
|
||
arrangement SEI messages
|
||
|
||
- H.265 encoder: added Screen Content Coding extensions support
|
||
|
||
- H.265 decoder: gained MAIN_444_12 profile support (decoded to
|
||
Y412_LE), and 4:2:2 12-bits support (decoded to Y212_LE)
|
||
|
||
- vaapipostproc: gained BT2020 color standard support
|
||
|
||
- vaapidecode: now generates caps templates dynamically at runtime in
|
||
order to advertise actually supported caps instead of all
|
||
theoretically supported caps.
|
||
|
||
- GST_VAAPI_DRM_DEVICE environment variable to force a specified DRM
|
||
device when a DRM display is used. It is ignored when other types of
|
||
displays are used. By default /dev/dri/renderD128 is used for DRM
|
||
display.
|
||
|
||
GStreamer OMX
|
||
|
||
- subframe support in H.264/H.265 decoders
|
||
|
||
GStreamer Editing Services and NLE
|
||
|
||
- framepositioner: new "operator" property to access blending modes in
|
||
the compositor
|
||
- timeline: Implement snapping to markers
|
||
- smart-mixer: Add support for d3d11compositor and glvideomixer
|
||
- titleclip: add "draw-shadow" child property
|
||
- ges:// URI support to define a timeline from a description.
|
||
- command-line-formatter
|
||
- Add track management to timeline description
|
||
- Add keyframe support
|
||
- ges-launch-1.0:
|
||
- Add an interactive mode where we can seek etc…
|
||
- Add option to forward tags
|
||
- Allow using an existing clip to determine the rendering format
|
||
(both topology and profile) via new --profile-from command line
|
||
argument.
|
||
- Fix static build
|
||
|
||
GStreamer validate
|
||
|
||
- report: Add a way to force backtraces on reports even if not a
|
||
critical issue (GST_VALIDATE_ISSUE_FLAGS_FORCE_BACKTRACE)
|
||
- Add a flag to gst_validate_replace_variables_in_string() allow
|
||
defining how to resolve variables in structs
|
||
- Add gst_validate_bin_monitor_get_scenario() to get the bin monitor
|
||
scenario, which is useful for applications that use Validate
|
||
directly.
|
||
- Add an expected-values parameter to wait, message-type=XX allowing
|
||
more precise filtering of the message we are waiting for.
|
||
- Add config file support: each test can now use a config file for the
|
||
given media file used to test.
|
||
- Add support to check properties of object properties
|
||
- scenario: Add an "action-done" signal to signal when an action is
|
||
done
|
||
- scenario: Add a "run-command" action type
|
||
- scenario: Allow forcing running action on idle from scenario file
|
||
- scenario: Allow iterating over arrays in foreach
|
||
- scenario: Rename ‘interlaced’ action to ‘non-blocking’
|
||
- scenario: Add a non-blocking flag to the wait signal
|
||
|
||
GStreamer Python Bindings
|
||
|
||
- Fixes for Python 3.10
|
||
- Various build fixes
|
||
- at least one known breaking change caused by g-i annotation changes
|
||
(see below)
|
||
|
||
GStreamer C# Bindings
|
||
|
||
- Fix GstDebugGraphDetails enum
|
||
- Updated to latest GtkSharp
|
||
- Updated to include GStreamer 1.20 API
|
||
|
||
GStreamer Rust Bindings and Rust Plugins
|
||
|
||
- The GStreamer Rust bindings are released separately with a different
|
||
release cadence that’s tied to gtk-rs, but the latest release has
|
||
already been updated for the upcoming new GStreamer 1.20 API (v1_20
|
||
feature).
|
||
|
||
- gst-plugins-rs, the module containing GStreamer plugins written in
|
||
Rust, has also seen lots of activity with many new elements and
|
||
plugins. See the New Elements section above for a list of new Rust
|
||
elements.
|
||
|
||
Build and Dependencies
|
||
|
||
- Meson 0.59 or newer is now required to build GStreamer.
|
||
|
||
- The GLib requirement has been bumped to GLib 2.56 or newer (from
|
||
March 2018).
|
||
|
||
- The wpe plugin now requires wpe >= 2.28 and wpebackend-fdo >= 1.8
|
||
|
||
- The souphttpsrc plugin is no longer linked against libsoup but
|
||
instead tries to pick up either libsoup2 or libsoup3 dynamically at
|
||
runtime. Distro packagers please ensure to add a dependency on one
|
||
of the libsoup runtimes to the gst-plugins-good package so that
|
||
there is actually a libsoup for the plugin to find!
|
||
|
||
Explicit opt-in required for build of certain plugins with (A)GPL dependencies
|
||
|
||
Some plugins have GPL- or AGPL-licensed dependencies and those plugins
|
||
will no longer be built by default unless you have explicitly opted in
|
||
to allow (A)GPL-licensed dependencies by passing -Dgpl=enabled to Meson,
|
||
even if the required dependencies are available.
|
||
|
||
See Building plugins with (A)GPL-licensed dependencies for more details
|
||
and a non-exhaustive list of plugins affected.
|
||
|
||
gst-build: replaced by mono repository
|
||
|
||
See mono repository section above and the GStreamer mono repository FAQ.
|
||
|
||
Cerbero
|
||
|
||
Cerbero is a meta build system used to build GStreamer plus dependencies
|
||
on platforms where dependencies are not readily available, such as
|
||
Windows, Android, iOS and macOS.
|
||
|
||
General Cerbero improvements
|
||
|
||
- Plugin removed: libvisual
|
||
- New plugins: rtpmanagerbad and rist
|
||
|
||
macOS / iOS specific Cerbero improvements
|
||
|
||
- XCode 12 support
|
||
- macOS OS release support is now future-proof, similar to iOS
|
||
- macOS Apple Silicon (ARM64) cross-compile support has been added,
|
||
including Universal binaries. There is a known bug regarding this on
|
||
ARM64.
|
||
- Running Cerbero itself on macOS Apple Silicon (ARM64) is currently
|
||
experimental and is known to have bugs
|
||
|
||
Windows specific Cerbero improvements
|
||
|
||
- Visual Studio 2022 support has been added
|
||
- bootstrap is faster since it requires building fewer build-tools
|
||
recipes on Windows
|
||
- package is faster due to better scheduling of recipe stages and
|
||
elimination of unnecessary autotools regeneration
|
||
- The following plugins are no longer built on Windows:
|
||
- a52dec (another decoder is still available in libav)
|
||
- dvdread
|
||
- resindvd
|
||
|
||
Windows MSI installer
|
||
|
||
- no major changes
|
||
|
||
Linux specific Cerbero improvements
|
||
|
||
- Fedora, Debian OS release support is now more future-proof
|
||
- Amazon Linux 2 support has been added
|
||
|
||
Android specific Cerbero improvements
|
||
|
||
- no major changes
|
||
|
||
Platform-specific changes and improvements
|
||
|
||
Android
|
||
|
||
- No major changes
|
||
|
||
macOS and iOS
|
||
|
||
- applemedia: add ProRes support to vtenc and vtdec
|
||
|
||
- The GStreamer.framework location is now relocatable and is not
|
||
required to be /Library/Frameworks/
|
||
|
||
- Cerbero now supports cross-compiling to macOS running on Apple
|
||
Silicon (ARM64), and Universal binaries are now available that can
|
||
be used on both X86_64 and ARM64 macOS.
|
||
|
||
Windows
|
||
|
||
- On Windows the high-resolution clock is enabled now in the
|
||
gst-launch-1.0 and gst-play-1.0 command line tools, which provides
|
||
better clock and timer performance on Windows, at the cost of higher
|
||
power consumption. By default, without the high-resolution clock
|
||
enabled, the timer precision on Windows is system-dependent and may
|
||
be as bad as 15ms which is not good enough for many multimedia
|
||
applications. Developers may want to do the same in their Windows
|
||
applications if they think it’s a good idea for their application
|
||
use case, and depending on the Windows version they target. This is
|
||
not done automatically by GStreamer because on older Windows
|
||
versions (pre-Windows 10) this affects a global Windows setting and
|
||
also there’s a power consumption vs. performance trade-off that may
|
||
differ from application to application.
|
||
|
||
- dxgiscreencapsrc now supports resolution changes
|
||
|
||
- The wasapi2 audio plugin was rewritten and now has a higher rank
|
||
than the old wasapi plugin since it has a number of additional
|
||
features such as automatic stream routing, and no
|
||
known-but-hard-to-fix issues. The plugin is always built if the
|
||
Windows 10 SDK is available now.
|
||
|
||
- The wasapi device providers now detect and notify dynamic device
|
||
additions/removals
|
||
|
||
- d3d11screencapturesrc: new desktop capture element, including
|
||
GstDeviceProvider implementation to enumerate/select target monitors
|
||
for capture.
|
||
|
||
- Direct3D11/DXVA decoder now supports AV1 and MPEG-2 codecs
|
||
(d3d11av1dec, d3d11mpeg2dec)
|
||
|
||
- VP9 decoding got more reliable and stable thanks to a newly written
|
||
codec parser
|
||
|
||
- Support for decoding interlaced H.264/AVC streams
|
||
|
||
- Hardware-accelerated video deinterlacing (d3d11deinterlace) and
|
||
video mixing (d3d11compositor)
|
||
|
||
- Video mixing with the Direct3D11 API (d3d11compositor)
|
||
|
||
- MediaFoundation API based hardware encoders gained the ability to
|
||
receive Direct3D11 textures as an input
|
||
|
||
- Seungha’s blog post “GStreamer ❤ Windows: A primer on the cool stuff
|
||
you’ll find in the 1.20 release” describes many of the
|
||
Windows-related improvements in more detail
|
||
|
||
Linux
|
||
|
||
- bluez: LDAC Bluetooth audio codec support in a2dpsink and avdtpsink,
|
||
as well as an LDAC RTP payloader (rtpldacpay) and an LDAC audio
|
||
encoder (ldacenc)
|
||
|
||
- kmssink: gained support for NV24, NV61, RGB16/BGR16 formats;
|
||
auto-detect NVIDIA Tegra driver
|
||
|
||
Documentation improvements
|
||
|
||
- hardware-accelerated GPU plugins will now no longer always list all
|
||
the element variants for all available GPUs, since those are
|
||
system-dependent and it’s confusing for users to see those in the
|
||
documentation just because the GStreamer developer who generated the
|
||
docs had multiple GPUs to play with at the time. Instead just show
|
||
the default elements.
|
||
|
||
Possibly Breaking and Other Noteworthy Behavioural Changes
|
||
|
||
- gst_parse_launch(), gst_parse_bin_from_description() and friends
|
||
will now error out when setting properties that don’t exist on
|
||
top-level bins. They were silently ignored before.
|
||
|
||
- The GstWebRTC library does not expose any objects anymore with
|
||
public fields. Instead properties have been added to replace that
|
||
functionality. If you are accessing such fields in your application,
|
||
switch to the corresponding properties.
|
||
|
||
- playbin and uridecodebin now emit the source-setup signal before the
|
||
element is added to the bin and linked so that the source element is
|
||
already configured before any scheduling query comes in, which is
|
||
useful for elements such as appsrc or giostreamsrc.
|
||
|
||
- The source element inside urisourcebin (used inside uridecodebin3
|
||
which is used inside playbin3) is no longer called "source". This
|
||
shouldn’t affect anyone hopefully, because there’s a "setup-source"
|
||
signal to configure the source element and no one should rely on
|
||
names of internal elements anyway.
|
||
|
||
- The vp8enc element now expects bps (bits per second) for the
|
||
"temporal-scalability-target-bitrate" property, which is consistent
|
||
with the "target-bitrate" property. Since additional configuration
|
||
is required with modern libvpx to make temporal scaling work anyway,
|
||
chances are that very few people will have been using this property
|
||
|
||
- vp8enc and vp9enc now default to “good quality” for the "deadline"
|
||
property rather then “best quality”. Having the deadline set to best
|
||
quality causes the encoder to be absurdly slow, most real-life users
|
||
will want the good quality tradeoff instead.
|
||
|
||
- The experimental GstTranscoder library API in gst-plugins-bad was
|
||
changed from a GObject signal-based notification mechanism to a
|
||
GstBus/message-based mechanism akin to GstPlayer/GstPlay.
|
||
|
||
- MPEG-TS SCTE-35 API: semantic change for SCTE-35 splice commands:
|
||
timestamps passed by the application should be in running time now,
|
||
since users of the API can’t really be expected to predict the local
|
||
PTS of the muxer.
|
||
|
||
- The GstContext used by souphttpsrc to share the session between
|
||
multiple element instances has changed. Previously it provided
|
||
direct access to the internal SoupSession object, now it only
|
||
provides access to an opaque, internal type. This change is
|
||
necessary because SoupSession is not thread-safe at all and can’t be
|
||
shared safely between arbitrary external code and souphttpsrc.
|
||
|
||
- Python bindings: GObject-introspection related Annotation fixes have
|
||
led to a case of a GstVideo.VideoInfo-related function signature
|
||
changing in the Python bindings (possibly one or two other cases
|
||
too). This is for a function that should never have been exposed in
|
||
the first place though, so the bindings are being updated to throw
|
||
an exception in that case, and the correct replacement API has been
|
||
added in form of an override.
|
||
|
||
Known Issues
|
||
|
||
- nothing in particular at this point (but also see possibly breaking
|
||
changes section above)
|
||
|
||
Contributors
|
||
|
||
Aaron Boxer, Adam Leppky, Adam Williamson, Alba Mendez, Alejandro
|
||
González, Aleksandr Slobodeniuk, Alexander Vandenbulcke, Alex Ashley,
|
||
Alicia Boya García, Andika Triwidada, Andoni Morales Alastruey, Andrew
|
||
Wesie, Andrey Moiseev, Antonio Ospite, Antonio Rojas, Arthur Crippa
|
||
Búrigo, Arun Raghavan, Ashley Brighthope, Axel Kellermann, Baek, Bastien
|
||
Nocera, Bastien Reboulet, Benjamin Gaignard, Bing Song, Binh Truong,
|
||
Biswapriyo Nath, Brad Hards, Brad Smith, Brady J. Garvin, Branko
|
||
Subasic, Camilo Celis Guzman, Chris Bass, ChrisDuncanAnyvision, Chris
|
||
White, Corentin Damman, Daniel Almeida, Daniel Knobe, Daniel Stone,
|
||
david, David Fernandez, David Keijser, David Phung, Devarsh Thakkar,
|
||
Dinesh Manajipet, Dmitry Samoylov, Dmitry Shusharin, Dominique Martinet,
|
||
Doug Nazar, Ederson de Souza, Edward Hervey, Emmanuel Gil Peyrot,
|
||
Enrique Ocaña González, Ezequiel Garcia, Fabian Orccon, Fabrice
|
||
Fontaine, Fernando Jimenez Moreno, Florian Karydes, Francisco Javier
|
||
Velázquez-García, François Laignel, Frederich Munch, Fredrik Pålsson,
|
||
George Kiagiadakis, Georg Lippitsch, Göran Jönsson, Guido Günther,
|
||
Guillaume Desmottes, Guiqin Zou, Haakon Sporsheim, Haelwenn (lanodan)
|
||
Monnier, Haihao Xiang, Haihua Hu, Havard Graff, He Junyan, Helmut
|
||
Januschka, Henry Wilkes, Hosang Lee, Hou Qi, Ignacio Casal Quinteiro,
|
||
Igor Kovalenko, Ilya Kreymer, Imanol Fernandez, Jacek Tomaszewski, Jade
|
||
Macho, Jakub Adam, Jakub Janků, Jan Alexander Steffens (heftig), Jan
|
||
Schmidt, Jason Carrete, Jason Pereira, Jay Douglass, Jeongki Kim, Jérôme
|
||
Laheurte, Jimmi Holst Christensen, Johan Sternerup, John Hassell, John
|
||
Lindgren, John-Mark Bell, Jonathan Matthew, Jordan Petridis, Jose
|
||
Quaresma, Julian Bouzas, Julien, Kai Uwe Broulik, Kasper Steensig
|
||
Jensen, Kellermann Axel, Kevin Song, Khem Raj, Knut Inge Hvidsten, Knut
|
||
Saastad, Kristofer Björkström, Lars Lundqvist, Lawrence Troup, Lim Siew
|
||
Hoon, Lucas Stach, Ludvig Rappe, Luis Paulo Fernandes de Barros, Luke
|
||
Yelavich, Mads Buvik Sandvei, Marc Leeman, Marco Felsch, Marek Vasut,
|
||
Marian Cichy, Marijn Suijten, Marius Vlad, Markus Ebner, Mart Raudsepp,
|
||
Matej Knopp, Mathieu Duponchelle, Matthew Waters, Matthieu De Beule,
|
||
Mengkejiergeli Ba, Michael de Gans, Michael Olbrich, Michael Tretter,
|
||
Michal Dzik, Miguel Paris, Mikhail Fludkov, mkba, Nazar Mokrynskyi,
|
||
Nicholas Jackson, Nicola Murino, Nicolas Dufresne, Niklas Hambüchen,
|
||
Nikolay Sivov, Nirbheek Chauhan, Olivier Blin, Olivier Crete, Olivier
|
||
Crête, Paul Goulpié, Per Förlin, Peter Boba, P H, Philippe Normand,
|
||
Philipp Zabel, Pieter Willem Jordaan, Piotrek Brzeziński, Rafał
|
||
Dzięgiel, Rafostar, raghavendra, Raghavendra, Raju Babannavar, Raleigh
|
||
Littles III, Randy Li, Randy Li (ayaka), Ratchanan Srirattanamet, Raul
|
||
Tambre, reed.lawrence, Ricky Tang, Robert Rosengren, Robert Swain, Robin
|
||
Burchell, Roman Sivriver, R S Nikhil Krishna, Ruben Gonzalez, Ruslan
|
||
Khamidullin, Sanchayan Maity, Scott Moreau, Sebastian Dröge, Sergei
|
||
Kovalev, Seungha Yang, Sid Sethupathi, sohwan.park, Sonny Piers, Staz M,
|
||
Stefan Brüns, Stéphane Cerveau, Stephan Hesse, Stian Selnes, Stirling
|
||
Westrup, Théo MAILLART, Thibault Saunier, Tim, Timo Wischer, Tim-Philipp
|
||
Müller, Tim Schneider, Tobias Ronge, Tom Schoonjans, Tulio Beloqui,
|
||
tyler-aicradle, U. Artie Eoff, Ung, Val Doroshchuk, VaL Doroshchuk,
|
||
Víctor Manuel Jáquez Leal, Vivek R, Vivia Nikolaidou, Vivienne
|
||
Watermeier, Vladimir Menshakov, Will Miller, Wim Taymans, Xabier
|
||
Rodriguez Calvar, Xavier Claessens, Xℹ Ruoyao, Yacine Bandou, Yinhang
|
||
Liu, youngh.lee, youngsoo.lee, yychao, Zebediah Figura, Zhang yuankun,
|
||
Zhang Yuankun, Zhao, Zhao Zhili, , Aleksandar Topic, Antonio Ospite,
|
||
Bastien Nocera, Benjamin Gaignard, Brad Hards, Carlos Falgueras García,
|
||
Célestin Marot, Corentin Damman, Corentin Noël, Daniel Almeida, Daniel
|
||
Knobe, Danny Smith, Dave Piché, Dmitry Osipenko, Fabrice Fontaine,
|
||
fjmax, Florian Zwoch, Guillaume Desmottes, Haihua Hu, Heinrich Kruger,
|
||
He Junyan, Jakub Adam, James Cowgill, Jan Alexander Steffens (heftig),
|
||
Jean Felder, Jeongki Kim, Jiri Uncovsky, Joe Todd, Jordan Petridis,
|
||
Krystian Wojtas, Marc-André Lureau, Marcin Kolny, Marc Leeman, Mark
|
||
Nauwelaerts, Martin Reboredo, Mathieu Duponchelle, Matthew Waters,
|
||
Mengkejiergeli Ba, Michael Gruner, Nicolas Dufresne, Nirbheek Chauhan,
|
||
Olivier Crête, Philippe Normand, Rafał Dzięgiel, Ralf Sippl, Robert
|
||
Mader, Sanchayan Maity, Sangchul Lee, Sebastian Dröge, Seungha Yang,
|
||
Stéphane Cerveau, Teh Yule Kim, Thibault Saunier, Thomas Klausner, Timo
|
||
Wischer, Tim-Philipp Müller, Tobias Reineke, Tomasz Andrzejak, Trung Do,
|
||
Tyler Compton, Ung, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Wim
|
||
Taymans, wngecn, Wonchul Lee, wuchang li, Xavier Claessens, Xi Ruoyao,
|
||
Yoshiharu Hirose, Zhao,
|
||
|
||
… and many others who have contributed bug reports, translations, sent
|
||
suggestions or helped testing.
|
||
|
||
Stable 1.20 branch
|
||
|
||
After the 1.20.0 release there will be several 1.20.x bug-fix releases
|
||
which will contain bug fixes which have been deemed suitable for a
|
||
stable branch, but no new features or intrusive changes will be added to
|
||
a bug-fix release usually. The 1.20.x bug-fix releases will be made from
|
||
the git 1.20 branch, which will be a stable branch.
|
||
|
||
1.20.0
|
||
|
||
1.20.0 was released on 3 February 2022.
|
||
|
||
Schedule for 1.22
|
||
|
||
Our next major feature release will be 1.22, and 1.21 will be the
|
||
unstable development version leading up to the stable 1.22 release. The
|
||
development of 1.21/1.22 will happen in the git main branch.
|
||
|
||
The plan for the 1.22 development cycle is yet to be confirmed. Assuming
|
||
no major project-wide reorganisations in the 1.22 cycle we might try and
|
||
aim for a release around August 2022.
|
||
|
||
1.22 will be backwards-compatible to the stable 1.20, 1.18, 1.16, 1.14,
|
||
1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
|
||
|
||
------------------------------------------------------------------------
|
||
|
||
These release notes have been prepared by Tim-Philipp Müller with
|
||
contributions from Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan,
|
||
Sebastian Dröge and Seungha Yang.
|
||
|
||
License: CC BY-SA 4.0
|